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Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any

On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv at mail.ru> wrote: > Hi all > > Have recently watched Matt Jordan's session on Kamailio World 2014 > > On slides 26-29 of his presentation > (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) > he speaks about a (completely...

Hi all We have recently upgraded Asterisk from 1.8.26 to 13.0.0 and now we have a whole lot of errors like this in our logs: [2015-01-05 06:43:36] ERROR[14636] stasis_cache.c: Attempting to remove an item from the SIP/ics0002-cached cache that isn't there: ast_endpoint_snapshot_type SIP/ics0002 Any ideas what it might be related to ? And how could we fix it ? Upgrade to 13.1.0 did not

Hi Marek Yes, here is a person with (mostly) working Asterisk 13 (chan_sip) + WebRTC (using sipml5 js lib) setup You can contact me directly, if you wish, I will try to help if I can As of the issue you have.. is it because you're working with FF 37 as browser ? I have not come across such issues since last summer, when FF (or Asterisk, don't remember exactly) had problems

Hi all I am using Asterisk 12.4.0 on debian 7.6 x64 I experience some troubles with some specific calls, so I want to dig into this as deep as possible I run these CLI commands: > sip set history on (answer: SIP History Recording Enabled) > sip set debug peer 501 (answer: SIP Debugging Enabled for IP: ...) > sip set debug peer 502 (answer: same with another IP) but when I

Hi Alexey, This is what works for me: [http.conf]: tlsenable=yes ; enable tls - default no. tlsbindaddr=144.x.y.z:8089 ; address and port to bind to - default is bindaddr and port 8089. tlscertfile=/etc/asterisk/keys/mycert.pem ; path to the certificate file (*.pem) only. tlsprivatekey=/etc/asterisk/keys/mycert.pem ; path to private key file (*.pem) only. Date: Tue, 13 Jan

Hi all ! I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in order to test WebRTC setup on my Asterisk PBX. I am using latest SVN version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677) If I make calls from softphones (Zoiper, X-Lite), which do not support DTLS at all, I can hear the Echo Test sound. BUT when I call from browser (I've tried latest Mozilla Firefox